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Well, it is still no problem to run miniSIPServer on this latest Ubuntu 16.10. Please enjoy it!
As we have said, since it is not a LTS version, if you deploy miniSIPServer in a business environment, it is better to keep your Ubuntu stay on version 16.04 or 14.04.
Some customers need invoice information. Most of them are on behalf of companies. To fit these requirements, we suggest you to purchase MSS through Avangate which is listed in our “buy now” page.
Please visit Avangate F.A.Q document:
http://www.avangate.com/shopper-support/shopper-faq.php
In this document, please refer to session “4. How do I receive the invoice?” for more details.
V28 is released today! We have spent several months on this version. The key feature of this version is “Lua service engine”.
As you know, previous MSS has a service engine which is written in Python language. It serves very well, but still has some limitations. Now the engine is rewritten in Lua language. Following items are some key points about why we make this decision.
(1) Lua is much simpler than Python. Python is full stack language,Lua is a embed script, it has less function but it is simple. We love Python but still find that Lua is suitable for MSS to provide service engine. We don’t need full functions, we only need the engine to packing MSS core functions and capabilities.
(2) The most important is stability. Python service engine uses one Python VM to serve all services. That means one unknown exception could block all services. With new Lua service engine, one Lua VM serves one service. That means one unknown exception can only block one service, other services can keep on working. That’s amazing, the whole system can step into higher level now!
(3) Faster! faster! faster! In fact, because of GIL, Python cannot provide high performance, so we have to limit the engine in service level. Lua doesn’t have such limitation, each Lua VM is tiny and independent, Now we can use it to be a service engine and will even use it to provide a basic call engine. The future is coming.
In V28, Python services are replaced with Lua services. You can find these Lua services in “lua/services” sub-directory. If you have modified Python services by yourself, you will have to update related Lua services.
Lua service engine is in background, you don’t need change any configuration by default.
One of our customers reported a problem that his external line was always offline with a voip provider. That’s very strange because “external line” is a very basic function of MSS and it works perfectly with lots of voip providers.
We captured the log and found the voip provider returned “400 Bad Request” message with following cause:
P-Registrar-Error: Invalid CSeq number
We checked the REGISTER messages, and think it is no problem in CSeq header. Following items are from MSS:
==> REGISTER sip:sip.xxx.com SIP/2.0 ... Call-ID: 18BF67854AE23D6D2CD772AFMSS002A0001. CSeq: 13 REGISTER ... <== SIP/2.0 401 Unauthorized ... Call-ID: 18BF67854AE23D6D2CD772AFMSS002A0001. CSeq: 13 REGISTER ... ==> REGISTER sip:sip.xxx.com SIP/2.0 ... Call-ID: 18BF67854AE23D6D2CD772AFMSS002A0001. CSeq: 14 REGISTER ... <== SIP/2.0 400 Bad Request ... Call-ID: 18BF67854AE23D6D2CD772AFMSS002A0001. CSeq: 14 REGISTER P-Registrar-Error: Invalid CSeq number ...
We checked RFC3261 to find “CSeq” in SIP-REGISTER procedures:
A UA MUST increment the CSeq value by one for each REGISTER request with the same Call-ID.
Obviously we are right. But why did peer side reject MSS’ messages?
Finally, we tried to send SIP-REGISTER with different ‘call-id’, and the problem was resolved! That made us confused again because in RFC3261 we can find the details of “call-id” in SIP-REGISTER procedures:
All registrations from a UAC SHOULD use the same Call-ID header field value for registrations sent to a particular registrar.
We think the voip provider is unprofessional. Unfortunally, it is hard for them to upgrade their system. So we have to add a switch varant to control MSS to fit this kind of situation.
[sip] gVarSipRegSameDialog=0
If you have the same problem with some voip providers, please add above parameter into “mss_var_param.ini” file and restart your MSS to enable it.
In the external line “outgoing calls” configuration, we add an item “no-answer timer”. This item is used to limit the no-answer timer value when make outgoing calls. Please refer to attached figure.
By default, its value is zero that means the line will use system default no-answer timer value. If it is set a value, the line will use it firstly.
For example, if it is set to be “15”, then the line will force to release the call in 15 seconds if peer side doesn’t answer the call.
We were busy on migrating our official website to new cloud computing systems in the past week. At the same time, we configure HTTPS for our website by default.
Now when you visit our website with HTTP connection, it will be converted to HTTPS automatically. It will make sure of communication between you and our website to anti invalid watching or modifications.
Since our system has been built on new cloud systems, it should be more stabler and faster. If you have any problem when visiting our website, please update us. We are appreciated for that.
The latest Ubuntu 16.04 was released yesterday. Since it is a LTS version, we downloaded it as soon as possible and made some test in lab.
Some libraries have been upgraded or changed in this version, we need update MSS to fit these modifications. If you want to try miniSIPServer on Ubuntu 16.04, or you want to upgrade your previous Ubuntu to 16.04, you need install latest miniSIPServer V27 (build 20160422) .
Please refer to attached figure, we run the latest MSS on Ubuntu 16.04.
One of our customers reported that his extensions have been cracked. We checked its MSS CDR records. It seems someone has cracked one extension’s password and used this extension number to make lots of calls.
Obveriously, it is a very dangerous problem. We think this “hacker” might send lots of SIP messages to MSS to try such extension’s password. MSS previous version doesn’t consider this scenario and always permit the SIP phone to keep trying its password until it is authorized.
To stop this, we upgrade V26 to support “fail to ban (F2B)” feature. Once SIP phone has failed to check authorization for several times in one minute, MSS will detect it as “scanning” and ban its IP address for several hours. All SIP messages from such address will be rejected directly. Then it is impossible for “hacker” to crack SIP passwords.
This feature is enabled by default and need configure nothing for it.
miniSIPServer V25 is updated to fix some bugs and refine system to be more stable. The most important change is that we cut ‘webRTC’ feature from this version and abover.
As we described in previous post, MSS webRTC feature can work with Chrome navigator. Chrome is upgraded to V48 and make some changes to webRTC and doesn’t consider compatibility with previous version. We think maybe webRTC is perfect for public network services, such as Google hangouts, but it is not suitable or flexible for small or middle size enterprise communication markets.
So we cut it from V25, and keep it in V24. If you are using webRTC feature, please keep your Chrome to V47 or lower versions.
Some virtual servers in cloud-MSS system have been changed, please pay attention to these items.
Each virtual SIP server will enable STUN feature. For example, if the SIP server address is “1234.s1.minisipserver.com”, its STUN server can also be the same address. That means “1234.s1.minisipserver.com” is also its STUN server address.
Now we suggest “stun.minisipserver.com” by default. It is a simple public STUN server for all virtual SIP servers. Of course, you can still configure your virtual SIP server address as your STUN server.
In voice-mail feature, we need a SMTP server to send emails with attached audio files. Each virtual SIP server can be configured with customers’ own SMTP servers. But we find it could make several problems. For example, most customers try to use Gmail SMTP server. Gmail SMTP server requires that you need enable POP/SMTP firstly, and grand other access. Most customers don’t know how to do that.
So we disable SMTP server configurations. All voice mails will be sent from our own SMTP server. Most important is that you will need check your spam box if you cannot find voice email in ‘inbox’.