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MyVoipApp documents and FAQ

External lines configurations

External lines configurations

We often configure miniSIPServer to connect VoIP carriers’ network with external lines. There are lots of VoIP carriers and someone always asks us how to configure external line.

In our step by step document, we give a demo to configure MSS to work with “call centric”. You can refer to this document for more details about VoIP networks and external lines. In another way, we give some more examples in chapter “External lines” of F.A.Q document. Please refer to these documents if you are interesting in it and hope they can be helpful to you.

https://www.myvoipapp.com/docs/faq/index.html

 

Relay media streams of SIP trunk outgoing calls

Relay media streams of SIP trunk outgoing calls

In some VoIP scenarios, we need configure “SIP trunk” to work with VoIP providers or gateways. When processing media streams, we hope (1) local users/phones should process their streams by themselves without MSS, and (2) MSS should help to relay media stream for all outgoing calls to peer SIP servers or gateways.

To fit these requirements, we update MSS V32 to be able to configure “relay media” item in “SIP trunk”. Please refer to following figure for more details.

Configure "relay media stream" item in SIP trunk outgoing call
Configure “relay media stream” item in SIP trunk outgoing call

By the way, MSS can only relay audio streams at this time, so video streams will be lost if you want to MSS to relay streams.

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
Concurrent calls of SIP trunk

Concurrent calls of SIP trunk

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.

Concurrent calls of SIP trunk
Concurrent calls of SIP trunk

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

Modification of “one number” service

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.

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.