Gilson – Page 8 – MYVOIPAPP products blog

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.

miniSIPServer on Ubuntu 16.10

2016/10/15 Gilson

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.

How do I receive invoice?

2016/10/08 Gilson

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 and Lua services

2016/09/16 Gilson

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.

Block anonymous calls

2016/08/27 Gilson

We can use “system black list” feature to block anonymous calls. Please click menu “services – system black list” and add a record:

caller number prefix = anonymous
called number prefix = *
rate = 100

This record means: 100% calls from caller “anonymous” should be blocked.

Citel Technologies, Inc. Announces Interoperability with MyVOIPApp miniSIPServer

2016/08/05 Gilson

AMHERST, NY and ShenZhen, P.R. China — August 4, 2016 — Citel Technologies, Inc., is pleased to announce that it has successfully completed interoperability testing with MyVoIPApp’s miniSIPServer, a software-based SIP PBX, designed for small and middle size companies, miniSIPServer is very easy to use with rich features and can work on multi-platforms, such as Windows and Linux whilst also fitting both IPv4 and IPv6 networks.

The Portico™ TVA™ offers companies the means to migrate their customers to VoIP without the unnecessary burden of ripping and replacing existing cabling infrastructure, purchasing new IP phones and installing Power over Ethernet switches. Customers can retain their existing digital, analog and Centrex phones without removing the existing switches by SIP enabling those phones through the use of the Portico™ TVA™. This is a quick and cost-effective means of VoIP migration.

“Used together, the two innovative solutions provide companies with the fastest and most cost-efficient means to upgrade from legacy systems to modern Unified Communications,” said Ian Gomm, VP of Sales & Marketing for Citel.

Citel and MyVoIPApp kicked off interop testing with miniSIPServer at the request of a high-profile customer who was looking to use this Windows-platformed softswitch for a cross-country network of customer service training centres. “We have undertaken interoperability with many IP PBXs over the years, and know that sometimes some systems are easier than others to complete. MiniSIPServer was new to us, but I was really pleased to hear engineers’ initial feedback that the system was easy to work with and interop had gone smoothly” said Andrew Davies, VP, Engineering at Citel. He continued “Then they showed me some quite sophisticated presence monitoring and Busy Lamp Field functionality and I saw the product was a great fit with the Portico™ TVA™, wherever digital business phones were the preferred handset. We look forward to taking the relationship further.”

About Citel Technologies, Inc. (citel.com)

Citel enables SMBs, large enterprises and service providers to realize the cost and productivity benefits of IP telephony while at the same time leveraging their existing PBX infrastructure. Businesses with single or distributed locations and PBX vendors can now deploy next-generation IP applications and services at their own pace, with minimal business disruption. Service providers can deploy Hosted IP telephony services quickly, without having to “rip and replace” existing enterprise PBX handsets and LAN cabling. Citel is based in Amherst, New York with offices in Loughborough, England (UK) and Toronto, Canada.

About MyVoIPApp (myvoipapp.com)

MyVoiPApp was founded in 2007, in ShenZhen, P.R. China. Although a small company its employees all have more than ten years experience in the field of communications and its focus on communication technology has facilitated strong growth and penetration in the VoIP marketplace.

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.

No-answer timer in external line

2016/05/16 Gilson

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.

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Software based SIP-PBX

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Author: Gilson