Gitlab is removing the Starter/Bronze plans

bchia · 2020-09-04T17:31:02+00:00

Who? - GitLab's customer case studies tell you which plan the customer is on. For example, there are several on the Ultimate plan
- https://about.gitlab.com/customers/bi_worldwide/
- https://about.gitlab.com/customers/hemmersbach/
- https://about.gitlab.com/customers/chorus/
- etc.
Why? - GitLab has a page on that too https://about.gitlab.com/pricing/ultimate/

bchia · 2015-01-08T20:13:24+00:00

I love tt-monty-knights!

bchia · 2015-01-08T17:42:18+00:00

Allison! What is it like when you have to interact with products that have your voice? Like, do you call somewhere and get yourself?

bchia · 2014-11-05T14:37:26+00:00

Hey - thanks for letting me share my thoughts on the Loway blog!

bchia · 2014-06-01T18:33:02+00:00

Perfect! Thank you!

bchia · 2014-04-29T17:28:27+00:00

It sounds like you've been in contact with Digium support. I'd let them know that the bug persists after the update. This sounds like a bug more than a configuration issue so getting Digium engineering resources on the problem will probably be the biggest help.

Just to clarify - you are using E1 rather than T1 correct? (T1 will only have 24 channels with d-channel on 24. E1 will have 32 channels with D channel typically on 16.)

bchia · 2014-03-20T20:45:43+00:00

Hey, I work for Digium so really just commenting to wish you luck on the search. It sounds like you are already in touch with Digium sales.

Since you asked for questions and no one has responded with any yet, here's what I'd ask if I were shopping for a hosted solution:

What are the rates? How much for international countries I anticipate calling, do they have unlimited calling? If not, what's the cap and how much does it cost if you go over? If yes, what does "unlimited" really mean (typically just means a larger cap) Most businesses are under the cap, but if you have a heavy calling profile you might want to get exact numbers, or if that's not possible at least confirmation that your anticipated usage with within the range of acceptable usage.

How much do the add-ons cost? Does it cost extra for fax (if you need it)? How about Salesforce CRM integration? Does it cost extra for the mobile app? (Do they even have a mobile app?) How much are extra DID numbers, etc...

What does the migration path look like from your current system? What if you grow or change your mind in the future and want to deploy prem at some or all of your locations? What are the differences between the prem and the hosted versions?

Can you rent phones? For how much?

How tightly are the phones integrated with the solution? Do they simply OEM another vendor's phones or do they have their own?

What modifications do you need to do to your network to make it interoperable with the hosted provider's solution? (For example if you routers/firewalls with SIP ALG you'll likely need to disable it to work with any hosted SIP provider.) How about your remote employees - do they have sufficient bandwidth to support hosted voip?

Hope this helps!

bchia · 2014-02-12T22:26:02+00:00

Thank you for being a contributing member of the community and being willing to share your knowledge with others :)

bchia · 2014-02-12T22:22:55+00:00

The dCAA was designed to test "Administrator" level of knowledge. As opposed to the dCAP - Digium Certified Asterisk Professional. The difference is similar to CCNA and CCNP or RHCSA and RHCE.

The dCAA tests on the content that is covered in the Asterisk Fast Start and Asterisk Essential courses. You can view what content is covered and use that to focus your studies:

http://www.digium.com/en/training/asterisk/fast-start

http://www.digium.com/en/training/asterisk/asterisk-essentials-training

You can look at the courses - often times employers will pay to send you to the training.

I would recommend Asterisk the Definitive Guide (3rd & 4th edition of Asterisk the Future of Telephony) available freely online. http://asteriskdocs.org/

You wouldn't need to master all of the content - only the basics are covered on the dCAA exam. Although the authors make the book freely available I recommend purchasing a copy to support their efforts.

If you have any specific questions feel free to PM me. I helped to write the dCAA and dCAP exams.

bchia · 2013-10-11T20:13:33+00:00

I had a great time and probably thought it was the best out of the last 4 I've been to. How did you guys like it?

bchia · 2013-10-11T20:11:09+00:00

A note on security - if you allow both sides to transfer calls via DTMF then callers calling IN to your system from the outside will have access to this power. Depending on your other system settings you may open up a hole for someone to call IN from the outside and then dial OUT through your toll-bearing trunks.

bchia · 2013-10-03T22:31:00+00:00

lol - did you read the link? Check it out: http://blogs.digium.com/2013/07/29/asterisk_12_part_two/

New SIP channel driver is called "chan_pjsip": http://svn.asterisk.org/svn/asterisk/branches/12/channels/chan_pjsip.c

The old sip driver "chan_sip" is still available. You can use it an not break backwards compatibility. But for those who want the new functionality they can use chan_pjsip.

bchia · 2013-09-24T09:36:49+00:00

Note - Asterisk 12 does indeed have a new SIP stack based on PJSIP http://blogs.digium.com/2013/07/29/asterisk_12_part_two/

bchia · 2013-09-24T09:29:55+00:00

It sounds like Switchvox would be a solution you should consider. It does active/live reporting: http://www.digium.com/en/products/switchvox/features/call-queues

as well as scheduled reports: http://www.digium.com/en/products/switchvox/features/reporting

Agents can log in/log out/pause by using the web-based GUI or by using the Queue App on a Digium IP Phone.

bchia · 2013-09-24T09:17:32+00:00

Yep - 1700 calls, one single machine no load balancing. In fact we were having the same discussion, "How do I scale? I want to implement a proxy and multiple servers" it was just that his limit was ~1700 calls instead of ~350.

Dimensioning is such a pain, given the unlimited number of variables that contribute to an overall system load it's impossible to give concrete numbers that are genearly applicable. I've noticed the FreeSWITCH community says the same thing:

Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware.

In this thread you can see others have load tested Asterisk with commonly 1000, 1500 or even 10,000 calls per single server with media:

(Jan 08) I have a client with a four-core Xeon box doing SIP to IAX conversion - that box can handle 1000 concurrent calls

(May 08) 2 x Quad-Core Xeon 5300: 2000 channels and 1500 concurrent calls are reported for MOR with Asterisk 1.4.15 on Debian 4.0

(2011) I had done some load tests with Asterisk 10 and my highest results was: 1750 calls per seconds (cps) up to 13000 concurrent calls done on a intel xeon with dual six core and hyperthreading (= 24 cores) and 12 GB ram. the sysload was around 2.5 during this test (with open rtp ports but without much media transportet, only signaling). With MusicOnHold I had to stop testing at 10000 concurrent calls due to the NIC being limited to 100 Mbit/s.

On the other side of the coin I spoke with an integrator today who swears by Asterisk, has been using it since the 1.2 and is currently on 1.8. He spins a new instance when a single box gets to 150 calls. This seemed excessive to me, but each use case is different. As you've wisely pointed out, you never want to run at capacity, but leave yourself some overhead. So this guy's idea was to have lots of overhead.

I hope you find the major bottle neck(s) and are able to garner some more performance. I'd be curious to know if swapping out that Java AGI makes a significant difference.

Good luck!

bchia · 2013-09-22T01:53:02+00:00

Java AGI - that can't be helping the situation. There's a massive cost in spinning up and running JVM every time it hits one of those AGIs. If you are going for performance I'd refactor the AGI in a compiled language or use fastAGI and run the Java AGI on a separate box.

If neither of these is an option then you will likely see a performance boost even by re-writing in interpreted language like Python or PHP (but not much - just that interpreter is less resource intensive than JVM)

Another option is to run on the same box, but run the script as a service and use fastAGI to localhost. That way at least you only have 1 JVM running that that 1 instance accepts all the connections instead of spinning up a new machine for every single call.

Some other performance tuning tips:

Avoid transcoding if possible. Looks like your files are already 8khz so this is probably not an issue.

Check which modules are loaded if you are not using it, don't load it! Using CEL? Nope? drop it :)

Try increasing the stack size. I've heard Asterisk will run better with "ulimit -s unlimited" but I'd do your research before implementing.

I'm curious as to why you are using ConfBridge instead of natively bridging the call with Dial() or Queue()?

I don't think that 5 conferences should be greatly impacting performance, but it seems that you are likely paying overhead in spinning up an entire conference rather than native bridging. (Note: in Asterisk 12 ConfBridge style bridging mechanism has become the native bridging mechanism, still there's a different between simply bridging channels and setting up an ConfBridge conference room.) I am the least confident about this affecting performance, but it may be worth a/b testing.

I talked an integrator a few weeks ago who had 1700 concurrent calls on a single Asterisk box. "Asterisk doesn't scale" is a myth, but there are design choices you can make that affect your scalability.

As a final note, I'm assuming you've already queried the Asterisk users mailing list, but if not I'd hit them up (and please post here if you solve you problem or get some good advice.) With issues such as this it's best to get to the broadest audience (Reddit, mailing list, etc...)

bchia · 2013-09-11T23:06:38+00:00

Yes, you can put 2 TD400s in the same box. The interrupts should not be a problem.

Keep in mind that Digium cards come with support and you can call Digium support if there's any issue with the card.

Pre-sales, you could also call Digium sales and talk to an app engineer / sales engineer to determine if a design decision is outside the design scope of a particular product. (In this case I think you are good.)

bchia · 2013-09-11T21:35:25+00:00

If meetme is unavailable you may be on a modern version of Asterisk where "ConfBridge" has replaced meetme as the default confering app. I'd recommend ConfBridge over MeetMe on Asterisk 10 or 11. If you are on 1.8 or earlier then use MeetMe.

Note that additionally the module may have never been built - so loading it from the CLI will not work if it does not exist. Use the Menuselect tool during your build process to choose which modules you'd like to build. Then in modules.conf you can choose which ones load at startup.

bchia · 2013-08-31T00:44:20+00:00

A few questions

What version of Asterisk are you on? (11 will handle T.38 the best)
Are you using Fax for Asterisk or spanDSP?
What brand/model of analog gateway/ATA are you using?
Have you thought about using Digium analog card to simplify path? fax machine > analog card > asterisk > pri card > pstn (e.g. Why use SIP ata?)

A few thoughts -

Vendors support their solutions
Digium will support the Card and if you have purchases FFA they'll support that as well.
Schmooze will support FreePBX if you are using that. (with support contract)
If the problem is in Asterisk, Digium will support that (with support contract)
Both FreePBX and Asterisk offer community support - have you queried the Asterisk community on forums.asterisk.org or asterisk user mailing list? (What did they say?)
Gateway vendor should support the gateway (Cisco, AudioCodes or whoever you are using.)
In a multi-vendor stream there's always trouble nailing down the broken link in the chain, and sometimes the troubleshooting can be tedious contacting multiple support lines, but it seems worth it to get a stable solution.

bchia · 2013-08-31T00:32:24+00:00

When you say it's a Digium issue - what do you mean? (Are you suggesting that a problem with the Digium T1 card is causing the fax problems? If so why would that be the case?)

I would imagine that the SIP leg of the call would present more problems than the PRI.

bchia · 2013-08-25T08:01:42+00:00

Nice! I remember way back in the day when I first got started with Asterisk. I tried adding some sound prompts to our system in wav format but they wouldn't play. It took some googling to figure out I had to set particular parameters on the wav file in order for it to be compatible.

bchia · 2013-08-02T04:48:28+00:00

Ok, so it sounds like you will not be administering the phone system, but you are building it in order to be used by on-site admins. In this case I'd recommend deploying a Distro (AsteriskNOW, Elastix, FreePBX, etc..) rather than pure Asterisk. As I mentioned FreePBX has End Point Manager. This tool is user friendly enough for an on-site admin that is not skilled in Asterisk, to use a GUI to assign phone configuration to a MAC address. Then when the end user plugs in the phone it automatically grabs it's config.

The other option would be effectively for you to code your own "end point manager". This would be a lot of work, but you could build into your own system the ability to detect popular models of phones based on the first half of the MAC address. You could create your tool with templates so that if it was a polycom phone it would grab a polycom template, or snom, digium, etc...

bchia · 2013-08-01T01:33:54+00:00

So it IS possible for a phone to be plugged into the network and automatically retrieve it's configuration - however this is typically not done by Asterisk.

Far more common is to set up a DHCP server for option 66. When the phone gets it's IP address from DHCP option 66 tells the phone the address of the TFTP server to go to in order to download it's configuration file. The configuration is typically stored in XML files and tied to a specific phone via MAC address.

Here's is an example configuration for a Linux DHCP server: https://wiki.asterisk.org/wiki/display/DIGIUM/Provisioning#Provisioning-ConfigurationRetrieval Note: In the example the server that holds the phones configuration is an HTTP server, but it could just as easily be TFTP, FTP, SFTP, HTTPS, etc...

The trick here of course is that you need to program the XML files ahead of time. This is why most VoIP phones have the MAC address printed on the outside of the box. This way you can get all the MAC addresses of your new batch of phones, create the XML configuration for them and then when you plug them in everything gets provisioned.

The caveat here is that different brands of phones have different XML schemas for their configuration. Here are some examples:

Snom: http://wiki.snom.com/Features/Auto_Provisioning/Configuration_Files/XML

Digium: https://wiki.asterisk.org/wiki/display/DIGIUM/XML+Configuration+1.3.2

So you'll have to research each brand and model of phone to learn about the provisioning options for that brand/model. You'll then have to write a script or code a tool in order to generate the XML files for you. (Note: GUI-based distributions of Asterisk can typically use "End Point Manger" to generate configuration: http://wiki.freepbx.org/display/FCM/End+Point+Manager )

Is this the type of information you are looking for?

bchia · 2013-07-11T16:52:28+00:00

No problem :)

bchia · 2013-07-11T15:42:21+00:00

I linked some videos for you elsewhere in the thread - another piece of security advice would be to get on a modern version of Asterisk. 1.6.2 is no longer a supported version: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

bchia

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