This post was mentioned on Twitter by asteriskbot: @NirSimionovich: Call Analytics – Beyond CDR analysis – Part I http://bit.ly/agwkl1 #Asterisk #Israel #opensource…

With businesses you must have a provider that can be responsible for a high level of service.

bbhenry – if you need any advice or help on this matter please contact me at moshe at flatplanetphone.com.

BTW – Voxbone is a reliable partner. Yes they do have their issues since at the end of the day they are just a clearing house for many carriers in different countries. BUT – they work to solve fast (in European terms…)
Bottom line, they are good.

Personally I’ll like to asked you something:
I’m working for a small company, who has decided after 3 years going to dCap cert and they choose me. Since I’m an employee, do you think this will be a positive or negative approach to get de cert?

I have previous experience with DIDX, and had been porting business numbers to them which didx claim that the numbers were hosted by their partner Global Crossing. And often times, we would loose incoming calls from didx passing to us. And didx would simply saying it’s global crossing’s server issue. And I don’t take full of what they tell me, because I know lot of times, IT company would just push the responsibility to another provider or partner they have(my previous company did that too). And it’s not a smart practice after all. Neglectful attitude will eventually drive your customers away and leave you bad word of mouth in the industry. And once you have bad reputation, it’s very hard to get up again. The company I previously worked for actually closed because of that.

Now I need to choose some good reliable outbound call provider that will pass caller ID along. Do you have any suggestions?

Sounds like your setup is fairly common from what I can tell. It is true that Heartbeat is mainly used to High-Availability. Load balancing requires the use of either OpenSER (or a similar software) or a vendor Session Border Controller. Bear in mind that the NAT based network environment of Amazon EC2 may pose some issues.
In terms of configurations, there is no single answer here, as it requires a proper analysis of your provisional business model. Feel free to contact me by email and I’ll see how I can assist you on your quest.

I’m not sure if you recall, but I’m well versed in GnuGK. Back in 2003 I used GnuGK+FreeRadius+MySQL to traverse over 25 Million minutes a month. I have to admit that the idea of running a cluster of GnuGK systems using Amazon EC2 is interesting, however, the nature of H323 and its incompatibility with NAT and the fact that Amazon EC2 is based on NAT networks, I would say that it poses an interesting issue.
In any case, GnuGK was optimally built to handle call setup requests, as it’s a GateKeeper, very much like a SIP proxy or SIP Session Border Controller, while Asterisk isn’t really built for that kind of function.

Here is the ultimate plan: To have an ATA device encode and decode calls using g729. Use openSer to direct traffic and handle registrations. So for instance if someone wanted to make a call from the ATA to the PSTN, OpenSer would help establish the invitation process between the ATA and pstn gateway and would not be included after that. Asterisk would only be used for voicemail and other features for the person on the ATA. I’m not sure if this is even possible put I would like for normal calls (from ATA to ..say someone on the PSTN) to not route their RTP traffic through Asterisk.

I’m sorry if this confusing. I’ve been furiously reading to try to understand but I’m sure I’m not close to being there. I would appreciate any help at all.
Thanks,
Blake McKeeby

Thanks and Regards
Rajib Deka

Well, if you copy pasted your AGI script as is, you have a small mistake with your ANSWER directive.

1. #!/usr/bin/php -q
2. answer();
10.
11. $agi->stream_file(”demo-congrats”,”#”);
12. do
13. {
14. $agi->stream_file(”enter-some-digits”,”#”);
15. $result = $agi->get_data(’beep’, 3000, 20);
$keys = $result['result'];
16.
17. $agi->stream_file(”you-entered”,”#”);
18. $agi->say_digits($keys);
19. } while($keys != ‘111′);
20. $agi->hangup();
21. ?>

In addition, it would appear that you didn’t even construct the AGI object correctly. If you refer to page 105 of my book, you would notice that lines 3 through 7 are missing from your script.

Try also to increase your console debug level via the /etc/asterisk/logger.conf file, that may shed some more information in regards to what breaks your script.

Cheers,
Nir

I have a problem with the following code (took from your book). It
doesnot make a beep sound and takes ainput. I am using Zoiper soft
phone to test it with an IAX2 account.
$result = agi->get_data(“beep”, 3000, 20);
say_digits() also not working.

My code looks like…

1. #!/usr/bin/php -q
2. answer();
10.
11. $agi->stream_file(“demo-congrats”,”#”);
12. do
13. {
14. $agi->stream_file(“enter-some-digits”,”#”);
15. $result = $agi->get_data(‘beep’, 3000, 20);
$keys = $result['result'];
16.
17. $agi->stream_file(“you-entered”,”#”);
18. $agi->say_digits($keys);
19. } while($keys != ‘111′);
20. $agi->hangup();
21. ?>

asterisk out put….

— Accepting AUTHENTICATED call from 127.0.0.1:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (),
> priority = mine
— Executing [999@agitest:1] AGI(“IAX2/rajib-14434″, “hello.php”) in
new stack
— Launched AGI Script /var/lib/asterisk/agi-bin/hello.php
AGI Tx >> agi_request: hello.php
AGI Tx >> agi_channel: IAX2/rajib-14434
AGI Tx >> agi_language: en
AGI Tx >> agi_type: IAX2
AGI Tx >> agi_uniqueid: 1234947670.52
AGI Tx >> agi_version: 1.6.0.5
AGI Tx >> agi_callerid: unknown
AGI Tx >> agi_calleridname: rajib
AGI Tx >> agi_callingpres: 1
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 16
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 999
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: agitest
AGI Tx >> agi_extension: 999
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> agi_threadid: -1217868912
AGI Tx >>
— AGI Script hello.php completed, returning 0
— Executing [999@agitest:2] Hangup(“IAX2/rajib-14434″, “”) in new stack
== Spawn extension (agitest, 999, 2) exited non-zero on ‘IAX2/rajib-14434′
— Hungup ‘IAX2/rajib-14434′

server configuration…..

Fedora core 8
asterisk server 1.6.0.5
phpagi-2.14

No other hardware is attached for PSTN line. I have tested the examples
with Zoiper.

Please help me out.

Thanks and Regards
Rajib Deka
rajib@excoflare.com