miniSipServer Cloud – Page 4 – MYVOIPAPP products blog

Concurrent calls of SIP trunk

2017/05/30 Gilson

By default, MSS previous versions don’t limit concurrent calls of SIP trunk. That means you can make or receive calls as much as you can. If peer sides don’t have enough resources, they will reject calls by themselves. But now in some scenarios, customers hope MSS can handle concurrent calls and limit them automatically.

To fit this requirement, we upgrade MSS to provide concurrent calls configurations in SIP trunk. Too much calls will be rejected by MSS itself. Please refer to following figure for more details about these items.

Please pay attention to these.

(1) These items are independent. You can configure different values for them to limit different concurrent calls for outgoing calls and incoming calls.

(2) If one of them is zero, in fact all them can be zero, that means only incoming calls can be received, or can only make outgoing calls outsides.

Modification of “one number” service

2017/05/03 Gilson

We upgraded miniSIPServer V30 today to change “one number, multi-devices” service in local user’s configuration. In previous versions, we don’t need configure anything to enable this feature in local user since it was enabled by default. Customers think it is good idea to reduce configuraiton workload, but it brings new management problem. In fact, they hope to be able to control which local users can have this feature. In most scenarios, only some local users have several phones with same number, others are not permit to do that.

To fit this requirement, we add a new optional item in local user’s configuration. Please refer to following figure for more details. By default, this service is not enabled now until you configure it obviously.

One number service right in local user's configuration. — One number service right in local user’s configuration.

This modification is applied to cloud MSS too.

Operator services

2016/11/07 Gilson

We added two traditional PBX services in new MSS version: operator break-in and operator overstep. Please refer to service document for more details.

Both local-MSS and cloud-MSS have been upgraded for these services. The latest MSS version is V29 now.

Next local MSS version will be V30 and we will focus on refining media gateway functions. If you have any suggestions or requirements, please update us and we are very glad to discuss with you.

Invalid CSeq number

2016/05/28 Gilson

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.

miniSIPServer updated and say goodbye to webRTC

2016/02/20 Gilson

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.

Refined SMTP library

2015/12/12 Gilson

In voice mail feature, MSS need use SMTP library to send emails. Since MSS can embed Python script functions, it is easy to use Python-smtplib to send email. That’s what we done and it works well, we are satisfied with it.

But smtplib is too old ( in python 2.7) to fit some modem SMTP servers’ requirements. It also has a shortage. It is synchronous. That means it can block thread when sending a email, then its performance is poor and cannot fit our requirements in cloud system.

Something is changed, we want MSS to be better, so we develop a new SMTP library to send voice mails. This SMTP library is asynchronous and can work perfectly with most SMTP servers. And, it is written in C/C++ language.

We upgraded MSS V23 and cloud-MSS to replace python-smtplib with this new SMTP library.

Hope you can enjoy latest versions.

By the way, since MSS V23 has been released for several months and we got very better result, we think it is time to release new LTS version which is V24 and new stable version which is V25 in the end of this year or in the beginning of next year.

miniSIPServer V21 released!

2015/05/26 Gilson

V21 is released today to support a new service: one number, multi-devices.

This service can enable MSS to accept several SIP phones with same local user’s number and authorization. And for incoming calls, all these phones will be ringing at the same time.

“One number” service looks like “ring group” feature. The difference is that “one number” service permits several SIP phones to share same local user profile, but “ring group” service requires SIP phones configured with different user profiles.

Please refer to following document for more details about this feature:

http://www.myvoipapp.com/docs/mss_services/one-number/index.html

By the way, Cloud-MSS has been upgraded to support this feature too.

Hope you can enjoy it!

start HTTPS

2015/05/05 Gilson

miniSIPServer.com is now HTTPS enabled. That means all your communication between your browser and our website will be encrypted.

Now you can visit cloud-MSS site with following linker:

https://minisipserver.com

DNS problem

2015/03/23 Gilson

Yesterday our data center has a DNS problem. In fact, it seems its DNS system was crashed. It made our servers failed to register all external lines. Then cloud system detected that all external lines were configured with wrong configurations. To avoid sending spam messages to peer sides or peer voip providers, our system will cut all “wrong” configurations automatically.

That’s too BAD news!

We promise this problem will not happen again. All our servers have been upgraded to enable several DNS systems including Google DNS.

We are very sorry for this problem. If your external lines are off, please configure them back manually.

Thank you for your patience and continued support.

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