Asterisk 1.4 (Final Release) Discussion

Hi All,

I want to make this Thread exclusive to Asterisk 1.4, Discuss the Problems you have discovered so far, and how you worked your way around them.

Sharing your story with everyone can be of great help to a lot of people, and your story does not have to be bad either, any good case studies would be interesting, as this can show that full testing has been conducted by you ina live environment or even a test lab.

For myself i have * 1.4 installed on a Dev Box in the office right here next to me :smiling_imp: I have hit a few hurdles as there are some changes between 1.2* and 1.4.

The things that i have noticed though with 1.4, and i am interested in seeing anyone else take on this;

  1. The SIP stack seems to be more efficient with version 1.4 (When you can get sessions to work properly)

  2. The DTMF handling also seems to be more efficient and is more responsive then the older stack, it also picks up the tones better as well, be it from a PSTN network or a Pure VoIP network.

  3. The new IAX2 also seems to be a better build, i notice with the Debug switched on you tend to see more information that is relevant that before, i could be mistaken, but it certainly seems to by far better then the previous version.

What i haven’t tested so far;

  1. The IVR system, i have not looked at what sample rates it can handle, what file formats are best suited for it, and how it generally handles everything over the previous versions of *. I would be interested to hear from others what challenges they have had to overcome, when they did get it to work the way they wanted it too, how it ended up playing out.

  2. One Asterisk box talking to another Asterisk box. I want to test the differences between IAX and SIP tunnels, i would like to gauge how each one of them fares up and which is best suited for the new version, due to the new stacks i would imagine each one would have some advances to them over the previous versions.

  3. CDR record Keeping. The older versions of Asterisk didn’t really have what i would call a good CDR system, as a result we just designed our own, but i would imagine that this new version would have some advancements too it, it may be one thing i might have to explore further one day.

What i would like to hear from other people;

  1. What do you use Asterisk for?

  2. If your Environment is Business only, then how many channels (IAX or SIP or even PSTN) do you have.

  3. How do you do your Interconnection? Is it over a Standard ADSL link? Are you located in a Co-Location/Telehousing site?

  4. What has been your major issue or sticking point since the migration or testing of the new version?

Don’t be shy, give your thoughts be it Negative or Positive, i am mainly doing this cause these forums don’t seem to cover this nature of Discussion, i only every see cries for help, but i never see any case studies, general discussions and or Sharing of Ideas and Solutions, which honestly is a crying shame :frowning:

I hope i can kick of the spirit of community sharing, sometimes just providing a very basic case study is a great help to many people, it can invoke ideas in people, and it can assist those who are working on maybe the same idea but are having troubles with certain sides of it.

My Case Study:

We are a Calling Card Company, we do Calling Cards and Call Back services, we use a Hybrid of VoIP and TDM. We purely use Asterisk only and basically threw out our CISCO/Nortel equipment in favor of Asterisk, this for me was a huge risk, but i held a lot of faith in Asterisk i honestly believe in its capabilities, whilst i understand it falls short on some things or compared to more traditional Telco Equipment, its flexibility is what lured me (OR Enticed me) to end up using it wholly and solely.

However Asterisk’s versatility is a major plus for it, the fact i can link up 10+ Asterisk Boxes via its IAX protocol over IPLC’s (International Private Leased Circuits), and keep deployment costs right down is a massive plus for it. Another striking thing about it, is, the fact that it can do Trans-coding of the same box, granted you build the right system for it (We do, our servers end up costing about $25KUSD) it ca handle everything with very little fuss, this is a very good plus for it, and is partly due to the reason to shift over to Asterisk.

Asterisk also exhibits strong voice quality attributes, i have heard many complaints about the voice quality Asterisk produces from_time_to_time, but i had to put that down to the system specifications based upon what work load they are putting Asterisk under, Yes Asterisk can run on very little specifications, however i knew that proper network planning would have to be done to ensure that you get the right mix.

Any way i can go on and on i supposed, but i am very much interested to hear about other peoples experiences.

NOTE*** If you are having troubles with doing something within Asterisk please don’t post it in here, i would like to keep this thread on topic all the way through it, if you have had problems and resolved them, then please do not hesitate to share your resolutions of the matter here.

Thank you all in advance for your contribution.



Ahhh Feel as though i am talking to myself but its all good :wink:

Anyway another problem detected which is a bit of a let down.

80% maybe more of the voice prompts on the Digium FTP download server, are pretty much screwed, some of the files do not start where they should and as a result you only seem to get the last part of the prompt.

This is somewhat of a let down, and quite saddening to see.

Oh well either we Hire someone to come in and record our own prompts, or we calculate the costs of paying Digium to do them, this we will leave up too the bean counters i suppose.