“Call forwarding” is a very traditional service in VoIP or communication fields. By default, SIP clients can send 3xx messages to miniSIPServer to invoke a forwarding. In another way, miniSIPServer can also directly invoke forwarding by itself.
But when the callee side is being forwarding, the caller side knows nothing about it. In most scenarios, the caller parties don’t care the forwarding. but some customers sometimes need to know what happens when the call is being forwarded.
miniSIPServer can send 181 “Call Is Being Forwarded” messages back to the caller side to update it that callee side is being forwarding. In the 181 messages, miniSIPServer will add a Call-Info header to indicate the forwarding information. Please refer to the figure below.
In this figure, there are two forwardings, (1) user B is being forwarded to user C; and (2) user C is being forwarded to user D.
The Call-Info header of the 181 message will indicate (1) the call is being forwarded, (2) who is being forwarded, and (3) who is being forwarded to. Please refer to the Call-Info header of the first 181 message which indicates user B is being forwarded to user C.
One of miniSIPServer cloud customers reported a bug that all his phones cannot register to the cloud system. We checked our networks and cloud nodes and found nothing.
We tried to capture SIP messages from his side but still got nothing. That means all SIP messages from his phones were lost, but his local network was OK, only SIP system was broken.
It is very strange. The customer finally found his local DNS was changed for unknown reasons. His local ISP returned wrong DNS records of our cloud system to his network. After changing the DNS server to Google DNS server, the problem was fixed and his VoIP network came back.
If all your SIP phones are offline and your network is confirmed to be ready, you can try to check DNS records. We suggest following tips to check the DNS records between Google DNS and your local ISP.
If you are working on windows system, you can use nslookup command to check DNS results. For example, we want to check the DNS result of virtual SIP server ‘1425.s1.minisipserver.com’ from Google DNS server which is ‘8.8.8.8’, we can use the command below.
nslookup 1425.s1.minisipserver.com 8.8.8.8
If you are working on Linux system, you can use dig command to check DNS result like following.
dig @8.8.8.8 1425.s1.minisipserver.com
You can check the DNS results from your local ISP’s DNS server. If its result is different with Google DNS result, that means your local ISP blocks our VoIP cloud system or its DNS results are contaminated for unknown reasons.
Personally, I suggest to use Google DNS server which is ‘8.8.8.8’ or cloudflare DNS server which is ‘1.1.1.1’.
By the way, Debian systems don’t have dig command by default. You need to install the dnsutils package to get such tool.
Debian 12 (bookworm) was released. It is the latest stable version and will be widely deployed in business environment absolutely. So we run and test the latest miniSIPServer on this system as usual. Of course, the result is perfect.
Please refer to the figure below.
If you want to build a VoIP system on Linux system, Debian 12 is a good choice.
Please refer to our online document for more details about how to install and run miniSIPServer on Debian systems. And I’m sure you’ll like the combination of Debian and miniSIPServer.
As we know, RFC3262 defines SIP reliability of provisional responses. It is an old feature and miniSIPServer ( both local versions and cloud versions) can support it for a long time. When working with traditional telcom carriers, this feature is mandatory, that means carriers will reject all incoming calls if they cannot support reliability of provisional responses.
RFC3262 defines a “100rel” parameter to indicate reliability of provisional responses, so we call it “100rel” capability. In normal, when invoking a call, the caller should make itself clearly that it can support “100rel” capability, and of course, the callee side should also make itself clearly that it requires to use “100rel” capability.
Basic call flow with 100rel
In the RFC3262, we can get following details:
…… the initial request contained a Supported or Require header field listing 100rel, and that there is a provisional response to be sent reliably. ……
UAS core … MUST contain a Require header field containing the option tag 100rel, and MUST include an RSeq header field.
Then both sides can establish reliability of provisional responses. Above figure describes the basic call flow. When UAC receives a 18x message which is a provisional response, UAC should send a PRACK message back to tell UAS that UAC has received its 18x message.
This is not a complex call procedure. We thought it wasn’t until several days ago. One of our cloud miniSIPServer customers reported he cannot make calls out. Then we traced his calls and get following call flow described below.
405 error in a 100rel procedure
Unbelievable …… this voip provider requires “100rel” in its 183 messages, but once miniSIPServer sends PRACK messages to confirm that, it returns “405 method not allowed” to reject them, and it causes every call failed.
Why?! If it cannot accept or support PRACK messages, why does it require “100rel” in its provisional responses?
It is quite easy to fix that. Just remove its “require 100rel” from 18x messages, miniSIPServer will not send PRACK messages back. But unfortunately, the team of this voip provider doesn’t know how to do that.
Our customer is blocked and his service has to be stopped. In another way, I personally think some VoIP providers use public open source servers, such as Asterisk, FreeSwitch, and so on, to build their solutions, maybe they don’t have enough experts to understand what they built.
So we update miniSIPServer to add an option in external lines configuration to disable reliability of provisional responses. Please refer to the figure below.
disable reliability of provisional responses
If you check this item, the INVITE messages sent from miniSIPServer will not have “support 100rel” parameter. Once you meet such a ridiculous VoIP provider, you can try this.
Ubuntu 22.04 (Jammy Jellyfish) is the latest LTS (long tem support) version, so it is quite important for miniSIPServer to support this version.
We try to install and run miniSIPServer on Ubuntu 22.04, it is very easy! And we are very glad that it is perfect to run miniSIPServer on it. Please refer to the figure below.
miniSIPServer on Ubuntu 22.04
If you are going to deploy a new VoIP network, it could be a good choice to run miniSIPServer on Ubuntu 22.04.
Debian 11 (Bullseys) was released at August 14th, 2021. It is the latest stable version and very important for business deployment. So we run miniSIPServer on this system and make some tests. The result is perfect!
It is no problem to run miniSIPServer on Debian 11. You can refer to online document for more details about how to run miniSIPServer on Linux systems.
In miniSIPServer, we can use IVR-XML script to enable our own services, such as automatic-attendant. With previous IVR-XML set, ‘callto’ action will invoke a call to destination and finish the whole IVR process.
But if we want to monitor some events in the call flow, such as we want to check ‘busy’ event and change the IVR flow to a new action, what should we do?
Now V37 is released and a key feature is updated in IVR-XML. We can use ‘monitor-events’ in ‘callto’ action to monitor some events and change the call flow if they are caused.
For example, the ‘callto’ action can be configured as below.
Above zip file is an example of new ‘callto’ action. You can save and unzip it into ‘xml’ sub-directory where miniSIPServer is installed and configure a new record to test it.
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
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 number service right in local user’s configuration.
