Asterisk - CSTA Siemens Openstage 40 problem: 489 Bad event

"Hi,Problem, Siemens Openstage 40 CSTA-XML vs Asterisk. When submitting a request via the context menu on the Openstage phone, Asterisk rejects the invite with an error. Is there a solution for Asterisk to process CSTA XML?"Asterisk version 16.28.0

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Asterisk has no support or knowledge of “CSTA-XML” itself.

I know that it doesn’t support CSTA itself, so what is the solution, please?

List from wireshark :62 9.530108 10.200.2.103 10.209.96.111 SIP 504 Status: 489 Bad event

I have no experience with CSTA-XML so I can’t offer anything further on immediate solutions. From an Asterisk perspective someone would have to write support for it.

Would upgrading to a higher version help solve this problem? Openstage from the context menu for requests, for example for forwarding - Content-Type: application/csta+xml, and the response from Asterisk is 489 bad event.

No version of Asterisk as provided by the project supports it. Upgrading Asterisk won’t make it appear.

Ok thank you

I started working with Asterisk only recently. I encountered a compatibility issue with the context menu of Siemens OpenStage 40 phones; all commands from the phone, such as call forwarding, call pickup, etc., are done via CSTA+XML, and Asterisk does not support this protocol. I’ve been tasked at work to transition to an open-source platform like Asterisk. Is there any chance or solution for this? Or would it be better to purchase phones that are fully compatible with Asterisk? Thank you.

I can’t comment on support by any other open source solution, for Asterisk my prior comments apply. Using a normal SIP phone would likely work fine.

The only “non-Asterisk compatible” VoIP phones on the market that have ever received significant development effort to make them compatible, have been the Cisco phones. I get periodically made fun of over on the FreePBX forums for continuing to push these phones - but it is a matter of numbers - Cisco shipped far more non-compliant VoIP SIP phones than any other phone maker in the world. As a result, there’s been significant effort making those phone compliant - and in fact - Cisco finally bowed to the inevitable and released firmware that makes many of their phones -completely- compliant with Asterisk so that is an option as well.

The Siemens phones are now $10USD per phone, see:

Wholesale Lot of 50 Siemens OpenStage 40 SIP HFA Phones with Handsets IP VoIP | eBay

They truly are junk, now, and the only people buying them anymore are supporting Siemens phone systems that are still alive.

If you need to replace a $10USD proprietary VoIP desk phone with a $10USD Asterisk-compliant VoIP equivalent, one of the best deals out there in the secondary (used) market IMHO are the Polycom phones - for example:

Lot Of 50 Polycom VVX 600 Gigabit IP Business Office Phones W Stands & Handsets | eBay

This is because Polycom was purchased by Hewlett Packard Enterprise - who frankly does not know what in the world to do with it. I have to wonder if this was one of those “CEO of Poly was golfing buddy of CEO of HPE who bailed out his board as a favor” deals. Polycom renamed themselves Poly but that didn’t help save them, and now HPE is sort of faffing around selling the latest Polycom phones in a sort of “throw it against the wall and see if it sticks” manner, but without a lot of heart in the effort.

In the meantime as people upgrade, there will be a lot of smarmy/lying/slick phone system salesguys who will be dissing “those old Polycoms” in an effort to capture a nice chunk of money and they will indeed succeed in the effort - causing tons of these phones to flood onto the secondary market.

And, they are very good phones, excellent sound quality, etc. They do have their weird provisioning quirks but nothing that hasn’t been documented, and they ARE supported by the latest OSS Endpoint Manager if you want to go 100% DIY. And OSS Endpoint Manager was recently worked on to get it to run under the latest php that FreePBX runs under. OSS Endpoint Manager is a darling favorite of Incredible PBX which is a kind of “wrapper” around FreePBX

I don’t know how many Siemens deskphones you have, but unless you are talking many thousands of deskphones - you could do exactly the same thing I did over the last year at my current employer which has around 300 extensions, all Cisco, and a year ago 3/4 of which were all really old Cisco that weren’t upgradable to standard-compliant phones. Just start slowly buying phones like the Polycoms off the secondary market and integrating them into your existing phone system, replacing the Siemens phones, in non-critical locations where people don’t need all of the features and the Siemens system can support them - then once you have collected all the phones you need, by then do a wholesale over-the-weekend transition of everything. I didn’t do a Cisco-to-Polycom migration I did a Cisco SCCP-to-Cisco Enterprise SIP in prep for a Cisco Enterprise SIP-to-Cisco-3PCC-SIP migration, but as you know the work is pulling and replacing phones, it’s not in the back end stuff that can be migrated in 30 minutes.

I completed my replacement project a few months ago and all Ciscos now are firmware upgradable from Cisco Enterprise firmware to Multiplatform firmware which is 100% compliant with Asterisk. All I need now, when I get ready to pull the trigger on the old UCM next year, is to find a phone consultant who is willing to migrate from the UCM to Asterisk, and we’ll do a prototype in the lab and then pull the trigger on the production system.

And if you want to get REALLY fancy, then just buy yourself a nice appliance:

Certified FreePBX Appliances | FreePBX - Let Freedom Ring

And you will get an open source PBX that’s fully supported by Sangoma.

and the Endpoint Manager that comes with the appliance will support out of the box those cheap Polycom VVX600’s. (although, that model is only tested with EPM not certified)

“Thank you for your response. In my network, there are 1,300 phones: 700 IP OpenStage phones and 500 analog phones connected through an analog Metriax gateway. The IP OpenStage phones are managed via DLS—templates, certificates, etc. The phones are currently registered to a Unify OpenScape PBX. I need to re-register the OpenStage phones to Asterisk. In my lab, I have set up Asterisk and registered the OpenStage phones, and everything is working except for the contextual menu, which communicates using the mentioned CSTA + XML, which is not supported by Asterisk. In the case of replacing the phones, I have an issue with phone management as I have the latest version of DLS for managing OpenStage phones. If I were to replace the IP phones, it would be quite costly, and I would lose the ability to manage the phones. The requirements for our telephony are basic calling, call pickup, call forwarding, and the use of the phone’s contextual menu and web interface by the users. We do not have any call centers or similar setups. The simplest solution would be a combination of OpenStage + XML connector + Asterisk. The migration from Siemens to Asterisk is driven by the need to save on data center costs.”

Correction, in your environment you have $13,000.00 USD worth of phones. :slight_smile:

Just because you have a lot of phones does not mean they are worth more than $10 ea. That’s a sunk cost fallacy argument. I’m NOT saying it is invalid for you - just that it is an invalid argument. In your case the labor for replacing 1300 phones is going to exceed the value of the phones, for sure, and I understand that. Remember, if you do nothing now, then you are just kicking the can down the road and 5 years from now you or someone else is probably going to be regretting you didn’t chuck them out when you had the chance.

In your shoes I STILL would consider a phone swapout - just over a longer time period. Like the old joke goes how do you eat an elephant - one bite at a time.

But in the meantime, if I was in your shoes, I would also seriously consider paying someone to code up an Asterisk module that handles CSTA+XML. It might be pretty quick and cheap to do. Ideally it could be open-sourced on github and help someone else with these phones. That would give you the breathing room to gradually migrate to a more supported phone so that 5 years from now when you want to use the latest Asterisk you aren’t stuck paying a developer again to brush up the coding on the CSTA module.