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Tips of using MyVoipApp products

Store your own audio files

Store your own audio files

miniSIPServer can support customers’ own audio files to replace default files. With previous MSS, customers have to backup and restore these files once they want to upgrade MSS.

This is a little trouble. With the new V32, we can resolve it now.

When MSS starts up, it will create a sub directory ‘cust_ann’ in ‘mss_ann’ directory, now all your own audio files can be stored in this directory. When MSS is uninstalled or upgraded, this directory and its files will not be deleted or replaced by default files, and MSS can get audio files from this directory directly when it starts up.

In windows system, it could be “d:/myvoipapp/minisipserver/mss_ann/cust_ann” directory by default. In Linux system, it could be “/opt/sipserver/mss_ann/cust_ann/”.

Please refer to our online document for more details about how to record own audio files.

https://www.myvoipapp.com/docs/others/how_to_record_your_own_audio/index.html

Connect to sonetel

Connect to sonetel

“sonetel.com” is a VoIP carrier who can provide local phone numbers. We can add “external line” to work with their servers. According to its description, there are some special items need to be cared:

  • Sonetel uses email address as SIP account and
  • It deploys SBC or proxy to process all incoming SIP messages.

Here we give a simplel example to describe how to work with Sonetel. We assume the SIP account is “abc@gmail.com”.

In MSS, please cilck menu “data > external line” to add a record.

Configure Sonetel lline
Configure Sonetel lline

In “Basic” tab, the line type should be “Connect to peer VoIP server”, the Account should be “abc” and the Domain is “gmail.com”.

By the way, the Password is the password you sign up in Sonetel, not your own email password.

Since all SIP messages are processed by sonetel SBC/Proxy, we need configure “outbound” information in the “Outgoing call” tab. Please refer to following figure.

Sonetel SBC
Sonetel SBC

The Sonetel proxy address is “sip.sonetel.com” which should be described in the email sent by sonetel.

Work with Chinese CTC IMS network

Work with Chinese CTC IMS network

Yesterday, we helped a Chinese customer to deploy MSS to work with CTC IMS network. In this scenario, CTC IMS network has ZTE soft-switch (according to User-Agent header in SIP messages) , we need be careful to cooperate with it.

Since CTC provides user name and password for authorization, we configure “external line” in MSS to do that. Following sections will illustrate some key points.

Authorization user name

By default, we often use “External line (account)” as authorization user name, but ZTE softswitch requires full URI format, so we need configure “The authorization ID should include address information” in external line. Please refer to following figure for more details.

Authorization user name
Authorization user name

For example, if this item is selected, the authorization name will be “+8612345678@gd.ctcims.cn” according to above figure.

If it is not full format, IMS network will return “403 Forbidden” messages to reject it. In fact, we think it is a bug in ZTE softswitch since there is “realm” and “domain” parameters in SIP authorization header. No matter the user name is full format or not, the device should pass it according to successful authorization itself.

Anyway, if you have same problem to cooperate with other IMS networks, please pay attention to it and configure such item to take a try.

Proxy

In Chinese CTC-IMS network, its “SIP server” is logic domain, not a real SIP device and cannot be visited. In above scenario, “gd.ctcims.cn” is its domain, not its real address. SIP messages should be routed to another device (we think it is a SBC or proxy), so we need configure “Via” address in MSS external line configuration. Please refer to following figure.

SIP proxy in IMS
SIP proxy in IMS
Block anonymous calls

Block anonymous calls

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.

Invalid CSeq number

Invalid CSeq number

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

No-answer timer in external line

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.

No-answer timer value
No-answer timer vaule in outgoing calls of external line.

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.

IP address authorization

IP address authorization

This feature was merged to the latest V25 (build 20160126).

Some special SIP devices, for example embeded devices in automaticated system, don’t have full SIP capabilities, they can make or receive simple SIP calls without account and password authorization. They even cannot send REGISTER messages to MSS to update their own status.

Yep, we can configure them as “SIP trunk” in MSS. but it will lost several key features, such as ringing-group. In some scenarios, customers hope to ring all such devices together, so we have to treat them as “local users”.

To fit these requirements, we add “IP address authorization” in local user’s configuration. That means MSS will not require SIP phones/devices to register them firstly, and will not check their account and password if their messages are from specific or configured IP addresses. Please refer to below figure for more details.

IP address authorization
IP address authorization

By the way, we update openAPI document according to the latest V25. If you are interesting in it, please refer to openAPI document.

SIP trunk between MSSes

SIP trunk between MSSes

1. Description

Some customers have several office branches and hope to establish VoIP connections between them. There are several methods to do that. Here we give an example to describe how to establish voip connections between two MSSes with SIP trunk feature.

2. Network topology

The network topology is simple. There are two office branches. Please refer to below figure.

Demo network
Demo network

Before we setup voip network, it is better to assign extension numbers. Different office numbers will effect how to configure call routing in both MSSes. In above figure, we can see the extensions in office 1 are 1xx, and extension numbers in office 2 are 2xx.

Both MSSes are configured with public IP address. If your MSS is behind NAT or router and you want to provide connection for outsides users, please refer to another document firstly.

Now we give detail configurations for it.

3. Configuration

In below configurations, items should be kept their default values if we don’t configure them obviously.

3.1 MSS1

Please click menu “data – SIP trunk”, then add a record:

SIP trunk ID = 1
Description = to MSS2
Server address = 10.23.x.x

Please click menu “dial plan – analyzed called number” , then add a record for routing 2xx to such SIP trunk.

called number prefix = 2
route type = SIP trunk
SIP trunk ID = 1

3.2 MSS2

It is almost same with MSS1.

Please click menu “data – SIP trunk” to add a record.

SIP trunk ID = 1
Description = to MSS1
Server address = 41.32.x.x

Please click menu “dial plan – analyzed called number” for routing calls to above SIP trunk.

called number prefix = 1
route type = SIP trunk
SIP trunk ID = 1
activate STUN in csipsimple

activate STUN in csipsimple

csipsimple is a very good sip client software in Android, we often suggest our customers to use it if they want to deploy voip network with Android phones.

As we know, there are always one-way or no-way audio problem. To solve this problem, STUN should be configured in softphone. But some customers often response it is hard to set STUN in csipsimple.

In fact, it is easy to do that. Please follow below steps which are described in csipsimple website too:

(1) Go on setting Settings > Network – Tick “Use Stun” and fill a stun server on the field bellow. If you are cloud-mss subscriber, you can use your virtual server address as your STUN server too. Of course, you can use csipsimple default STUN server.

(2) You can also try to use ICE in addition to STUN if STUN alone doesn’t solve the problem : Settings > Network – Tick “Use ICE”.

Refine called number

Refine called number

V14.4 is updated to support a new feature in “dial plan” process. This feature is “refine called number”.

“Refine called number” can be used to refine called number before calls are routed to external lines or SIP trunks. It is the last chance to change called number to fit different requirements from peer VoIP servers.

For example, one of our customer has two VOIP accounts. One is from local provider, another is from international provider. These two VoIP providers have different number format requirements, and our customer only want to has one kind of dial plan for both of them. So we can configure “refine called number” to refine the final destination number to fit it. This scenario is illustrated below.

Scenario

As described above, there are two VoIP accounts, and users need dial “90xxxx” to make outbound calls. “9” is MSS default outgoing call prefix. “0” is required by local VoIP provider. At the same time, the international VoIP provider requires that the number format should be “0086xxxx”.

After compare these number formats, we can find that we only need change prefix “0” to “0086” for international VoIP account.

Step 1: configure an independent “outgoing group ID” for international VoIP account

Please click menu “Data / External line” and select the account to edit, then please click “Outgoing call” tab and configure following item:

Outgoing group ID = 1

Step 2: configure “number transition”

We need configure a new record to change preifx “0” to “0086”. Please click menu “Dial plan / Transition” to add a new record:

Transition ID = 1
Transition type = Replace
Start position = 0
Length = 1
Replace string = 0086

Step 3: refine called number for specific outgoing group

Please click menu “Dial plan / Refine called number” to add a new record:

Outgoing group ID = 1 <== defined in step 1
Called number prefix = 0
Transition ID = 1 <== defined in step 2

Here we maybe have a problem: the called number prefix is “0”, why? why not analyze “9” prefix? It is because that “9” has been deleted in “analyze called number” procedure and the number has been changed to “0xxxx” before it is sent to external line or SIP trunk, so we should analyze prefix “0” to refine final called number.