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.
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.
For example, if this item is selected, the authorization name will be “+firstname.lastname@example.org” 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.
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.
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.
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 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:
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.
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.
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.